SIP Trunking with Interoute One (e-office)

Interoute is one of our partners in SIP services and a Microsoft Gold Partner. A couple of year's ago e-office connected LCS2005 to the VoIP cloud of Interoute. However, our current infrastructure only supports Microsoft Office Communications Server 2007 R2, so we needed to make sure that Interoute’s service will also work correctly with the OCS R2 Mediation Server.

On the 16th of October our e-office Microsoft Team connected to Interoute via SIP trunking (using SIP over TCP, as supported in OCS 2007 R2) and of course federation. Together with Interoute we accomplish some tests to get certified.

Since then Interoute has achieved its certification for SIP Trunking Services Qualified for Microsoft Office Communications Server 2007 R2

What is Interoute One?

Interoute One is a unique development brought to you by Interoute, the largest pan-European VoIP network owner. It is designed to integrate seamlessly with Microsoft OCS.

Interoute is the VoIP provider to many large Corporates and currently carries over 6 billion VoIP minutes per year.

Main features:

Feature:

Business quality VoIP

Free calls between your sites on the Interoute network

Free calls to other Interoute One customers

Realise real cost savings with our low cost calls package

Reduce your mobile roaming costs

No new hardware required – easy integration

Traditional architecture:

cons:

Complex

Expensive

Requires greater expertise

Geographically limited

Interoute architecture:

pros:

Simple online setup

Zero cost

Auto configured

So what are the steps required to try it out for yourself… before eBay-ing your PBX?

Basically you need to do the following steps:

Step 1.

To request a trial go to on www.InterouteOne.com and click ‘click here to apply’.

Fill out the request form, making sure you to enter your corporate email address. Webmail addresses like yahoo, gmail or hotmail are not eligible. Link Sing-Up for FREE!

Note: no inbound numbers are not available with the trial. This feature is only for postpaid customers.

Step 2.

Once your request will be approved, you will receive an email from Interoute with a link directing you to the sign-up page.

Domain Name: This is the 'Sip domain' that your users are configured to use. If the user name is like user@company.com, the domain name is company.com. Most of the time this domain name is the same as the corporate SMTP domain space.

Make sure an OCS 2007 R2 Mediation Server is installed and joined to the domain for example (contoso.local). A Mediation Server must be able to pass SIP requests and media between the Enterprise Voice infrastructure and a media gateway connected to the PSTN.

Assign this OCS 2007 R2 Mediation Server with one NIC and assign two separate IP addresses. Configure one IP address to listening for Office Communication Server traffic. Configure the other NIC for listening for voice gateway traffic. There can only be one default gateway!

Sometimes:

The customer may have decided to implement their OCS 2007 R2 Mediation Server on a machine that has two interfaces, where they may dedicate one interface for OCS 2007 R2 facing communication and the other for external communication. Unlike the access edge proxy, the OCS 2007 R2 Mediation Server may not always intelligently select the right interface for packets destined for the Interoute One service.

Be careful when using a static NAT mapping set up, this may cause issues on the interconnect (traffic sentfrom the other interface may end up originating from a different public IP, which will be dropped by the Interoute servers). If the customer can see SIP traffic leaving the OCS 2007 R2 Mediation Server, but it doesn't seem to arrive at Interoute Servers, please check that the traffic is using the correct interface.

Make sure your internal pool name is filled in (example: pool1.contoso.local) and make also sure that the PSTN next hop address is correct. In your case the PSTN next hop address is the public IP address of the Interoute One servers (89.*) this information fill be provided by Interoute in case you have any questions about that.

Step 5.

Make sure within the OCS 2007 R2 Management Console that only users prepared as Enterprise Voice users are checked with the Enterprise Voice option and that the line URI is filled in. Example (+312......) You can configure Enterprise Voice on per user basis.

Step 6.

Make sure all users who are using Enterprise Voice have installed Microsoft Office Communicator 2007 R2 (MOC2007) minimal version 3.5.6907.

Step 7.

Make sure numbers entered in Microsoft Office Communicator 2007 R2 can normalized to E.164. The format of the telephone number associated with the UC-enabled user is E.164 (example: +123456789). Therefore, if a user enters a number that is in different format (example: extension 1234) it has to be manipulated into the E.164 format.

OCS 2007 R2 will take this number and search the corporate directory to find the user who has a matching number and then voice mail will be routed to the correct user. OCS 2007 R2 uses normalization rules to translate these number formats and uses an internal translation service to perform canonicalization transformation. In our case we use only one normalization rule.

Check your Route within OCS 2007 R2. When Communications Server determines that a dialed number needs to be routed to a PSTN gateway, the routing table is queried to determine the optimal gateway for the call. In our case the route is (.*) because we only have 1 normalization rule. So there is no need for logic routing.

Step 11.

Open ports on your FW (only outbound) only to the specific public IP addresses of Interoute (89.*) When configuring the OCS 2007 R2 Mediation Server , you are advised to accept the default media port gateway range of 60,000 to 64,000. The default range media port range enables the server to handle up to 1,000 simultaneous voice calls. We as e-office expanded this range.

There you go! Outbound calling will go through the services of Interoute One. Inbound calling will be forwarded to your OCS 2007 R2 infrastructure. All: cheap, easy to use and just perfect! Sign it up for free and do not hesitate to contact me if you have any questions.

The benefits of SIP Trunking are outstanding. This system helps in real time communication, cost effective service, great quality voice, and reliable service. Through one line, one can accomplish all types of work. Thanks